A current high level of interest exists for integrating more function into common telephone systems. Advanced function telephones are viewed as requiring the ability to transmit both voice and data. In general, a new telephone terminal concept, well known in the industry, includes a telephone handset, a keyboard and a character display, at a minimum. Voice signals are quantized within the telephone instrument and encoded as a 64 kilobit per second pulse code modulation (PCM) signal stream by the telephone handset circuits. Transmission over a communications link to and from such a terminal is all in the form of digital patterns which may be encoded into an analog form for transmission if desired. A key problem in designing such a terminal is the design of the integration mechanism and technique for accommodating both digitized voice signals and digital data signals. Integration is necessary both from a functional and transmission viewpoint. Integration for transmission is desirable for office voice terminals because it simplifies the transmission scheme and makes the integration of data with the voice stream easier. However, integration is mandatory for a subscriber data terminal if it is to be applied to the communications link through the telephone system.
One such subscriber terminal has been described by Toru et al in an article entitled, "An Approach to Multiservice Subscriber Loop System using Packatized Voice/Data Terminals", appearing in the International Symposium on Subscriber Loops and Services, March, 1978, conference record, page 161. In this reference, one design for a future telephone terminal is set forth.
In the referenced article, the design taken is that for the digital voice signal to be integrated with the other terminal data sources by using packetized transmission techniques. Packetizing makes the voice signal compatible with the data transmission which is in packets. Also, packetizing the voice signal allows excess channel capacitiy available during silence gaps in the speech stream to be utilized for data transmission. The general approach described has been widely explored for optimizing transmission performance in integrated voice and data systems.
Equally well explored in the prior art are the problems of voice packet transmission. These problems include delay due to node buffering, controlled packet ordering, multipath timing jitter and other technical problems well known to those of skill in the art. The necessity of solving these problems creates a level of complexity and cost which is inconsistent with the basic objective of low cost professional or subscriber terminal service.